| 00:00.14 | *** join/#asterisk elzid (~IceChat7@91.107.38.125) |
| 00:00.41 | elzid | hello all - have a desperate question RE AGI |
| 00:01.48 | elzid | Im executing a read application command to read dtmf into a variable and I cant get the variable back for some reason no matter what I try! I've done a NoOp on the var in extensions.conf and have verified that it has been set by the read |
| 00:01.59 | elzid | anyone can help me please? |
| 00:02.05 | ritzt3ch | INbound Call coming in FXS And it takes about 3 Seconds to finally kick in the Dialplan |
| 00:02.28 | ritzt3ch | Starting simple switch Dahdi7-1 Then 3 - 4 seconds later it rings |
| 00:03.18 | elzid | Ive tried: ast_exec("GET VARIABLE cont1"); |
| 00:03.23 | elzid | but it returns empty |
| 00:05.02 | [TK]D-Fender | spaceout: The DAHDI ones clearly. |
| 00:05.25 | [TK]D-Fender | elzid: PASTEBIN is your friend <- |
| 00:05.27 | [TK]D-Fender | ~pb |
| 00:05.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:06.02 | [TK]D-Fender | ritzt3ch: You too |
| 00:09.14 | elzid | TKD-Fender: http://pastebin.com/tWqP7Shd |
| 00:09.27 | elzid | TKD-Fender: What are your thoughts? |
| 00:10.29 | pabelanger | in sip.conf, is regexten=516@context a valid setting? |
| 00:11.07 | elzid | TKD-Fender: this is clearer: http://pastebin.com/hLiKWpzc |
| 00:11.11 | ritzt3ch | i did a tail on the /var/log/asterisk/full and this is what i get .... |
| 00:11.12 | ritzt3ch | [2010-10-03 16:45:29] VERBOSE[17847] chan_dahdi.c: -- Starting simple switch on 'DAHDI/8-1' |
| 00:11.12 | ritzt3ch | [2010-10-03 16:45:33] WARNING[17847] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (No such device) |
| 00:11.22 | ritzt3ch | notice the 4 second differece |
| 00:11.23 | spaceout | [TK]D-Fender: /etc/dahdi/modules http://pastebin.ca/1953705 |
| 00:12.12 | [TK]D-Fender | spaceout: chan_dahdi.conf <---- |
| 00:12.25 | spaceout | kk |
| 00:12.28 | spaceout | http://pastebin.ca/1953706 |
| 00:12.33 | spaceout | that's the dahdi-channels.conf |
| 00:12.38 | [TK]D-Fender | ritzt3ch: what card, * ver, etc? |
| 00:12.42 | spaceout | getting chan_dahdi.conf right now |
| 00:13.28 | [TK]D-Fender | pabelanger: No. |
| 00:13.38 | [TK]D-Fender | pabelanger: there is a separate regcontext |
| 00:13.40 | ritzt3ch | TDM400P REV I Board 5 (no Echo Canceler) |
| 00:14.07 | [TK]D-Fender | ritzt3ch: Something sounds b0rked |
| 00:15.29 | elzid | [TK]D-Fender: Any thoughts please? |
| 00:15.48 | spaceout | [TK]D-Fender: http://pastebin.ca/1953709 |
| 00:15.50 | ritzt3ch | well in chan_dahdi.c i saw matchdigittimeout=3000 ? /usr/src/asterisk/channels/chan_dahdi.c |
| 00:15.58 | ritzt3ch | 3 seconds ? |
| 00:16.40 | ritzt3ch | CID is fine i even manually set it in chan_dahdi.conf |
| 00:17.00 | [TK]D-Fender | elzid: Code AND call output with AGI debug |
| 00:17.18 | WIMPy | ritzt3ch: Could it be, you have some ambiguity in your dilplan? Sounds like that might be your problem. |
| 00:17.25 | [TK]D-Fender | ritzt3ch: That is on the FXS side (IN from FXS)? |
| 00:18.02 | pabelanger | [TK]D-Fender: got it working. Had to prepend my context: regcontext=sipregistrations&context |
| 00:18.32 | [TK]D-Fender | spaceout: where is the CALL to look at? |
| 00:18.51 | spaceout | [TK]D-Fender: ? |
| 00:19.26 | [TK]D-Fender | spaceout: Show us a call coming IN that doesn't get answered |
| 00:19.40 | spaceout | asterisk doesn't generate anything, the RX bar in dahdi_monitor goes up on each ring |
| 00:23.32 | spaceout | [TK]D-Fender: http://pastebin.ca/1953710 |
| 00:23.41 | [TK]D-Fender | spaceout: O RLY? NOTHING? verbose 10 and nothing? pastebin "dahdi show channels", "dahdi show channel 1", "dahdi show status". |
| 00:23.46 | [TK]D-Fender | 1 pastebin. |
| 00:24.37 | [TK]D-Fender | elzid: Also waht ver of *? |
| 00:25.20 | spaceout | http://pastebin.ca/1953712 |
| 00:27.34 | *** join/#asterisk digilink (~digilink@tn-76-5-159-171.sta.embarqhsd.net) |
| 00:30.07 | [TK]D-Fender | spaceout: Indeed no channel 1 there... |
| 00:30.20 | [TK]D-Fender | spaceout: what does "dahdi_cfg -vvv" at CLI give you? |
| 00:30.29 | elzid | TKD-Fender: 1.4 |
| 00:30.56 | elzid | TKDeFender: Asterisk 1.4.30 built by root @ nexus on a i686 running Linux on 2010-04-17 15:32:41 UTC |
| 00:31.47 | spaceout | [TK]D-Fender: http://pastebin.ca/1953715 |
| 00:31.50 | spaceout | oops, i'm a noob |
| 00:31.58 | spaceout | let me fix that real quick |
| 00:34.55 | ritzt3ch | IN From FXO to FXS but its not even TOUCHING my dialplan YET (i built a Duplicate box With everything the same and SIP trunk inbound was INSTANT) |
| 00:35.32 | spaceout | [TK]D-Fender: after i fixed the fxs/fxo line in the conf i had fucked up |
| 00:35.34 | spaceout | http://pastebin.ca/1953721 |
| 00:35.49 | ritzt3ch | i had a test lab of the paging server at my office and then its now at the customers site and the only thing is they they complain about a 3 second lag (which i think is fine but to them they are picky as hell) |
| 00:36.56 | spaceout | [TK]D-Fender: and from dmesg http://pastebin.ca/1953725 |
| 00:37.33 | WIMPy | ritzt3ch: That freaks out most users. |
| 00:38.17 | [TK]D-Fender | spaceout: wcfxo: WARNING: Error inserting zaptel (/lib/modules/2.6.18-92.1.22.el5/misc/zaptel.ko): Device or resource busy FATAL: Error inserting wcfxo (/lib/modules/2.6.18-92.1.22.el5/misc/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 00:38.23 | [TK]D-Fender | spaceout: ZAPTEL? Pardon?! |
| 00:38.41 | [TK]D-Fender | spaceout: WTF is ZAPTEL doing in there when you're supposed to be using DAHDI? |
| 00:38.59 | spaceout | idk |
| 00:39.26 | [TK]D-Fender | spaceout: What have you installed, and how? |
| 00:40.03 | spaceout | compiled zaptel 1.4 from source a while back |
| 00:40.34 | spaceout | it sounded likea good idea at the time |
| 00:40.46 | spaceout | card wasn't working with dahdi then, either |
| 00:40.48 | elzid | TKD-Fender: more info as requested: http://pastebin.com/1umNivfp |
| 00:44.02 | [TK]D-Fender | spaceout: scripts sound screwed up |
| 00:44.45 | spaceout | [TK]D-Fender: suggestion? |
| 00:45.02 | WIMPy | ritzt3ch: Maybe you should take a look at your dialplan instead of your dahdi config. |
| 00:45.07 | [TK]D-Fender | elzid: [Oct 4 01:32:31] ERROR[6241]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe <-- boatloads of AGI erros due to inputting and outputting when you shouldn't be |
| 00:46.09 | elzid | TKD-Fender: I couldn't find any reason/solution for it - but everything (except the topic) is working despite these - clearing these was my second objective |
| 00:48.05 | [TK]D-Fender | elzid: -- User entered '2' <-- also this looks like your AGI DID get the variable back |
| 00:48.51 | elzid | TKD-Fender: that's the default Read app displaying user input - but ive not been able to request this into a variable |
| 00:48.52 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 00:48.56 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 00:49.18 | elzid | TKD-Fender: the application detects and stores the dtmf - but I just cant get it back into a variable! |
| 00:49.34 | elzid | TKD-Fender: in AGI |
| 00:49.36 | [TK]D-Fender | elzid: Well your AGI handling code seems quite broken hence those erros everywhere |
| 00:50.27 | elzid | TKD-Fender: Could you cast your eyes over my ast_get_var funct - what am I doing wrong there? |
| 00:50.32 | *** join/#asterisk spaceout (spaceout@gateway/shell/xzibition.com/x-sqgryjmrofkimzye) |
| 00:53.08 | [TK]D-Fender | elFirst you aren't running your PHP in QUIET MODE which every doc tells you you have to do. |
| 00:53.11 | [TK]D-Fender | elzid: ^ |
| 00:54.03 | elzid | TKD-Fender: sorry, I was at one point then Ive gone through absolutely everything throughout the day to solve issues |
| 00:54.06 | WIMPy | throws in a penny towards a new tab key for [TK]D-Fender ;-) |
| 00:54.42 | [TK]D-Fender | WIMPy: I missed the "z" that would have made a match... |
| 00:55.16 | WIMPy | Ok, then a new PC speaker that tells you it didn't match. |
| 00:56.56 | [TK]D-Fender | WIMPy: I'm using a GUI desktop here with a sound card... which is busy streaming me The Daily Show right now thank you :p |
| 00:57.22 | [TK]D-Fender | WIMPy: Please to be not interrupting the Jon Stewart, kplzthxbibi |
| 00:57.23 | *** join/#asterisk spaceout (spaceout@gateway/shell/xzibition.com/x-yewzpkqdrdodcokh) |
| 00:57.33 | *** part/#asterisk fofware (~Fabian@186.124.144.194) |
| 00:57.52 | WIMPy | See, so the PC Speaker would be the right direction. |
| 00:58.38 | [TK]D-Fender | thinks about bludgeoning WIMPy with his PC speaker... but then upgrades to his 300W 4x12 speaker cab.... |
| 00:59.44 | spaceout | [TK]D-Fender: what do you suggest i do at this point? |
| 01:00.37 | [TK]D-Fender | spaceout: What do you have running on that server right now? |
| 01:00.52 | WIMPy | also has 350 or was it 325W speakers, but not connected. This room is a little too small. |
| 01:01.39 | spaceout | [TK]D-Fender: asterisk 1.6, freepbx 2.8 |
| 01:02.09 | [TK]D-Fender | spaceout: Anything else? |
| 01:03.00 | [TK]D-Fender | WIMPy: http://montreal.kijiji.ca/c-buy-and-sell-musical-instruments-amps-pedals-Line6-Spider-II-HD150-Half-Stack-W0QQAdIdZ232700448 <----- this |
| 01:03.57 | WIMPy | Nice one. |
| 01:04.21 | spaceout | [TK]D-Fender: just the required shit for freepbx |
| 01:06.04 | WIMPy | 's actually got 35 cm, but what good is it without the right room? |
| 01:08.30 | [TK]D-Fender | WIMPy: Mine was for live shows... |
| 01:08.45 | [TK]D-Fender | WIMPy: Never actually got any use like that which is why I'm flipping it. |
| 01:08.59 | [TK]D-Fender | spaceout: modprobe wcfxo |
| 01:09.17 | [TK]D-Fender | spaceout: then do "dahdi_cfv -vvvv". Pastebin them both |
| 01:10.03 | WIMPy | The big speakers were nice in the big room I had before, but now I have small rooms which are good to store a pile of rdundant technology. |
| 01:10.23 | WIMPy | Guess it was better before... |
| 01:13.09 | [TK]D-Fender | WIMPy: When this is sold I'm left with a single 75W 12" amp, 1 florrboard multi-effects processor, 1 table-top analog synth, and my Roland XP-30 workstation... |
| 01:13.32 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 01:13.41 | [TK]D-Fender | WIMPy: I used to basically run EVERYTHING through a mixer and single set of 2x15W studio monitors. |
| 01:13.49 | [TK]D-Fender | WIMPy: including my PC. |
| 01:15.07 | WIMPy | Actually I have no Idea what the speakers I use now are capable of. But I managed not to damage them so far, even tho connected to that 600W amplifier. |
| 01:16.47 | WIMPy | I don't even know where they came from. They must be pretty old, but actually the sound is very good so I don't miss the big ones. |
| 01:16.58 | spaceout | [TK]D-Fender: http://pastebin.ca/1953740 |
| 01:17.12 | WIMPy | Probably from a time when quality still existed. |
| 01:18.01 | [TK]D-Fender | WIMPy: my 30W worth of studio monitors is more then planty for my room and everything through 1 mixer leading to it.... Everything is is for live performance... can't wait till I ditch that stack I linked to you |
| 01:18.31 | [TK]D-Fender | spaceout: pastebin "dmesg" |
| 01:19.16 | spaceout | [TK]D-Fender: http://pastebin.ca/1953742 |
| 01:19.26 | [TK]D-Fender | spaceout: root@pbx:~ $ modprobe wcfxo ---- WARNING: Error inserting zaptel (/lib/modules/2.6.18-92.1.22.el5/misc/zaptel.ko): Device or resource busy <-- is * still RUNNING? |
| 01:22.36 | spaceout | [TK]D-Fender: http://pastebin.ca/1953745 |
| 01:23.25 | spaceout | same dmesg |
| 01:23.48 | [TK]D-Fender | spaceout: Ok, stop your server. Reboot. pastebin dmesg again. |
| 01:28.45 | spaceout | http://pastebin.ca/1953749 |
| 01:35.40 | [TK]D-Fender | Zaptel Version: 1.4.12.1 |
| 01:35.56 | [TK]D-Fender | spaceout: We should not be seeing zaptel here. it does not work with * 1.6 |
| 01:36.26 | spaceout | [TK]D-Fender: i just removed, reinstalled dahdi from source |
| 01:37.37 | [TK]D-Fender | spaceout: This begs to differ. You need to kill of the kernel modules, etc from Zaptel. You have conflicting stuff in there |
| 01:37.56 | spaceout | did an rm on all zaptel modules |
| 01:38.16 | [TK]D-Fender | spaceout: Clearly not |
| 01:39.41 | spaceout | [TK]D-Fender: http://pastebin.ca/1953752 |
| 01:40.32 | spaceout | [TK]D-Fender: dmesg http://pastebin.ca/1953753 |
| 01:41.19 | [TK]D-Fender | spaceout: I see a whack of unknown symbol errors. This makes it look like your kernel got updated and devalidated all of your kernel modules |
| 01:41.31 | [TK]D-Fender | spaceout: recompile & install DAHDI |
| 01:42.17 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 01:45.21 | *** join/#asterisk breardo (~no@234-200-29-134.hcc.mnscu.edu) |
| 01:46.49 | spaceout | [TK]D-Fender: done |
| 01:47.56 | spaceout | zaptel still appears to be there |
| 01:52.18 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:56.59 | spaceout | [TK]D-Fender: i think i'm making progress |
| 01:57.10 | spaceout | http://pastebin.ca/1953755 |
| 01:57.23 | *** join/#asterisk kink (~kink@184.52.124.226) |
| 01:58.54 | spaceout | [TK]D-Fender: updated http://pastebin.ca/1953757 |
| 02:00.24 | spaceout | [TK]D-Fender: asterisk outputs http://pastebin.ca/1953758 |
| 02:01.36 | spaceout | [TK]D-Fender: now does it just need configuration? |
| 02:07.23 | [TK]D-Fender | spaceout: I still see no channel in * |
| 02:07.37 | [TK]D-Fender | spaceout: And no output from dahdi_cfg that doesn't end in tragic error |
| 02:08.12 | [TK]D-Fender | spaceout: I would recommend you go follow some comprehensive PIAF upgrade guide or download a more appropriate base to start from |
| 02:19.56 | *** join/#asterisk LoRd_RaHl (~LoRd_RaHl@c-68-62-0-30.hsd1.mi.comcast.net) |
| 02:22.03 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 02:22.25 | elzid | Hi - can anyone help me use php to extract the 2 from between the paranthesis at the end of the below please? |
| 02:22.27 | elzid | 200 result=1 (2) |
| 02:22.50 | elzid | a regex maybe or a gawk statement? but need to run from within php |
| 02:23.19 | elzid | its a response to an agi get variable request |
| 02:24.37 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 02:26.00 | elzid | any help is appreciated |
| 02:28.10 | *** join/#asterisk LoRd_RaHl (~LoRd_RaHl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 02:28.34 | LoRd_RaHl | any one good with Polycom digitmaps? |
| 02:38.37 | [TK]D-Fender | elPerhaps you shouldn't be building your own AGI framework yourself. You'd probably be better off using a pre-made one. |
| 02:40.32 | elzid | ive tried premade ones but am not getting any values back - I can see the Get Variable command returning the value in the above form but premade functions are not capturing the data part within () - I'd like to see if I can capture it but not sure how to in php - do you have any idea how to parse that string? |
| 02:41.05 | bougyman | elzid: php5 -r 'preg_match("/\((\d+)\)$/", "200 result=1 (2)", $matches);echo $matches[1];' |
| 02:41.08 | bougyman | 2 |
| 02:41.21 | bougyman | probably not the best way to handle that. |
| 02:41.25 | elzid | bougyman - many thanks - ill give that a shot now |
| 02:41.27 | bougyman | looks like a line protocol. |
| 02:45.34 | *** join/#asterisk coppice (~chatzilla@m121-202-12-226.smartone-vodafone.com) |
| 02:45.52 | elzid | bougyman: didnt work for me im afraid, if I want to capture the following output, how can I? |
| 02:45.53 | elzid | AGI Rx << GET VARIABLE cont |
| 02:45.53 | elzid | AGI Tx >> 200 result=1 (5) |
| 02:46.15 | elzid | the Rx is from this request: $call->execute_agi("GET VARIABLE cont"); |
| 02:46.43 | [TK]D-Fender | elzid: I doubt he looked back to read all of your code.... |
| 02:46.49 | bougyman | i'm concused at what you want. |
| 02:46.59 | bougyman | waht is the string you receive? |
| 02:47.15 | bougyman | just "200 result=1 (5)" ? |
| 02:47.32 | elzid | bougyman: according to agi debug it is what u say |
| 02:47.37 | bougyman | php5 -r 'preg_match("/^(\d+)\s(.*?)\s\((\d+)\)$/", "200 result=1 (5)", $matches);var_dump($matches);' |
| 02:47.43 | bougyman | run that on your command line |
| 02:47.50 | bougyman | is that what you are expecting? |
| 02:48.02 | elzid | I have to put it in the script so I tried this: |
| 02:48.05 | elzid | $kh=preg_match("/\((\d+)\)$/", $call->execute_agi("GET VARIABLE cont"), $matches);echo $matches[1]; |
| 02:48.07 | Juggie | i've never seen agi return a var in brackets before |
| 02:48.26 | bougyman | i don't see anything in brackets. |
| 02:48.32 | elzid | juggie: the brackets is for the data |
| 02:49.11 | bougyman | elzid: why don't you capture $call->execute_agi in to a variable and take a look at it? |
| 02:49.15 | bougyman | it may not be what you think it is. |
| 02:49.51 | elzid | its always empty - it returns an array of some sort |
| 02:50.39 | [TK]D-Fender | elred_: Again, this is YOUR function. Who sees how screwed up your OTHER code is? They aren't scrolling that far back. |
| 02:51.18 | elzid | this is the response I get when I try to capture to a variable AGI Rx << VERBOSE "Array" |
| 02:59.17 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 03:00.41 | bougyman | seems like it's not returning what you expect. |
| 03:01.51 | elzid | bougyman: that's clearly saying its an array data type but any idea how I can just extract the 5 that's within () into a scalar var? |
| 03:02.07 | *** join/#asterisk timeshell_atwork (~chatzilla@206.248.136.108) |
| 03:02.25 | bougyman | that makes no sense. |
| 03:02.29 | bougyman | i cannot respond. |
| 03:02.32 | timeshell_atwork | Greetings. Any advice on blocking incoming "friendly-scanner"s? |
| 03:02.35 | elzid | ok thanks |
| 03:02.39 | bougyman | it's an array but theres a (5)? |
| 03:02.53 | ChannelZ | timeshell_atwork: a firewall |
| 03:02.57 | bougyman | timeshell_atwork: fail2ban |
| 03:03.13 | ChannelZ | or an AR-15 if you know where they live |
| 03:03.15 | timeshell_atwork | any bfd rules that have been made to block it? |
| 03:03.29 | bougyman | iptables -P INPUT DROP |
| 03:04.55 | timeshell_atwork | That's not what I menat |
| 03:05.27 | timeshell_atwork | I already DROP that way. However, I don't know how to make a bfd rule that detects the friendly-scanner messages |
| 03:05.28 | WIMPy | Pulling the cable also works if you prefere it more rustic. |
| 03:19.29 | Dovid | i believe i have seen the same name over and over I think it's called sipfriendly or something of the sort. you can try searching based on that |
| 03:35.27 | drmessano | oh god |
| 03:35.28 | drmessano | http://www.xkcd.com/797/ |
| 03:35.33 | drmessano | ^^^^ New fav XKCD |
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| 04:17.41 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 04:18.59 | dandate2 | so i'm an american out here in the philippines (god help me in dealing with these people). the major internet provider is installing me a T1 line and tells me that my american standard cisco sa8000 router does not have direct serial port connectivity to their european standard E1, and thus i have to pay more per month to have their router on deck. is this true? |
| 04:22.05 | WIMPy | Check the manual? |
| 04:25.18 | *** join/#asterisk voipuser (~FreePBX01@ool-1859807a.static.optonline.net) |
| 04:30.26 | voipuser | what is the magic number supposed to be in dahdi_monitor when dialing in to an asterisk Milliwatt() application? |
| 04:40.47 | ChannelZ | http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html |
| 04:45.55 | voipuser | ChannelZ thanks. I read that, and get 14844 when dialing a telco milliwatt, however the asterisk Milliwatt application seems to be giving only about 4600 even on the tx side (which should be unaffected by gains and line loss because it is before it even leaves the asterisk box) |
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| 04:57.33 | kaldemar | dandate2: "serial port connectivity from router to E1"? that doesn't make much sense, but sounds like they are not installing a T1 for you, but an E1, and your router only does T1. |
| 05:13.00 | voipuser | to answer my own question above about milliwatt. some serious searchnig found this: https://issues.asterisk.org/view.php?id=15386 |
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| 05:33.16 | *** part/#asterisk voipuser (~FreePBX01@ool-1859807a.static.optonline.net) |
| 05:44.31 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-kzjkettubabtmciw) |
| 05:48.18 | *** join/#asterisk bluOxigen (ssf@unaffiliated/bluOxigen) |
| 05:49.39 | dandate2 | well i told the ISP that my american cisco router has a 50 pin serial port to connet to PRI, they told me they are on european standard |
| 05:49.49 | dandate2 | and insisted i pay an extra $50/mo to rent their router |
| 05:50.00 | dandate2 | which isnt a lot to me, but these philippinos would sell their soul for $50 |
| 05:50.07 | dandate2 | which is why im wondering if they were just outright lying to me |
| 05:50.11 | WIMPy | Maybe you should ask them what kind of interface they deliver? |
| 05:50.17 | WIMPy | Or what options they offer? |
| 05:50.57 | dandate2 | they said its E1 connectivity because they dont offer T1 but in the american frame of mind its basically the same |
| 05:51.14 | WIMPy | And don't call it PRI. It's not a phone line. |
| 05:51.47 | WIMPy | Well, the bitrate differs. |
| 05:52.04 | WIMPy | But most equipment should be fine with either rate, I guess. |
| 05:52.17 | dandate2 | thats what im wondering |
| 05:52.26 | dandate2 | if i find this serial port connects to whatevers coming out the wall imma be pissed heh |
| 06:03.01 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
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| 06:27.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.137.173) |
| 06:37.49 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 06:37.51 | schmidts | good morngin |
| 06:40.35 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:41.33 | wdoekes2 | morning |
| 06:43.09 | schmidts | hello walter |
| 06:43.11 | schmidts | how are you? |
| 06:43.35 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.177.137.173) |
| 06:43.57 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.137.173) |
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| 06:51.17 | kaldemar | dandate2: you don't connect to a T1 or E1 with a serial port. |
| 06:52.00 | WIMPy | What kind of interface do you expect then? |
| 07:00.04 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:09.55 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:11.02 | sbingner | dandate2, most cisco routers will do either E1 or T1 on the same port AFAIK |
| 07:12.17 | sbingner | but SA8000 is a cable box.... |
| 07:15.08 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 07:31.13 | wdoekes2 | just fine schmidts, thanks |
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| 07:40.46 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-201-110.pools.arcor-ip.net) |
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| 07:50.06 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 07:52.15 | verywiseman | when i hang up call , asterisk continue executing context steps , so if i redial number , it is busy , shortly when i hang up ,asterisk don't hang up also until it go to Hangup(),why? |
| 07:55.48 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
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| 08:01.24 | *** join/#asterisk superciuc (~Alexandru@ip62.trivenet.it) |
| 08:02.27 | *** join/#asterisk BANSAL (~bansal@117.207.81.52) |
| 08:02.55 | superciuc | Hello! I have asterisk at 155% CPU, one dead channel (a call that has 77 hrs) and the console full of lines like: |
| 08:02.58 | superciuc | [Oct 4 09:28:57] DEBUG[32309]: channel.c:1217 channel_find_locked: Avoiding deadlock for channel '0x7f2a840799f0' |
| 08:02.58 | superciuc | [Oct 4 09:28:57] DEBUG[32267]: channel.c:1222 channel_find_locked: Failure, could not lock '0x7f2a840799f0' after 199 retries! |
| 08:03.09 | superciuc | is there any way to resolve the issue without restarting asterisk? |
| 08:03.20 | superciuc | soft hangup on that channel doesn't work (I'm using * 1.6.0.22 on a debian 5.0.3) |
| 08:10.50 | *** join/#asterisk ramindia (~currys@9.96.63.202.southernonline.net) |
| 08:11.06 | ramindia | chan_dahdi.c:2790 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 08:11.29 | ramindia | i have issue with with E1 pri,. its going down and up |
| 08:11.42 | ramindia | Status: Provisioned, Down, Active |
| 08:12.53 | verywiseman | when i hang up call , asterisk continue executing context steps , so if i redial number , it is busy , shortly when i hang up ,asterisk don't hang up also until it go to Hangup(),why? |
| 08:14.18 | ChannelZ | verywiseman: are you using the 'g' flag in your Dial statement(s)? |
| 08:15.30 | verywiseman | ChannelZ, no |
| 08:16.31 | verywiseman | ChannelZ, is it necessary to put g? |
| 08:16.41 | ChannelZ | no |
| 08:17.12 | ChannelZ | When the channel terminates it should stop executing contexts so something is overriding that. Show some console output of this happening |
| 08:17.22 | schmidts | verywiseman do you have a FXO phone in use? |
| 08:18.08 | verywiseman | no, i have ip phone |
| 08:19.21 | verywiseman | ChannelZ, with g it terminate context if i hang up , is this right? |
| 08:20.30 | ChannelZ | no with 'g' it continues on |
| 08:22.49 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:23.40 | ChannelZ | RE: show verbose console output on a complete session from dialing to hangup to whatever it happening afterwards |
| 08:30.13 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 08:30.37 | ChannelZ | ...or don't. I'm going to bed |
| 08:31.58 | schmidts | gn8 channelz ;) |
| 08:32.45 | *** join/#asterisk coppice (~chatzilla@m121-203-215-12.smartone-vodafone.com) |
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| 08:50.22 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 08:51.55 | angryuser | Good day, asterisk 1.6.0.20, my console is spammed by this messages "<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/00085D10C9B7-000005ea]" |
| 08:52.39 | angryuser | Any idea to what it is related ? |
| 08:53.31 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com) |
| 08:55.08 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 08:56.36 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 08:57.36 | schmidts | maybe a malformed keep alive message |
| 09:03.23 | WIMPy | Or just an empty one? |
| 09:04.55 | schmidts | angryuser turn on sip debug for this peer and see if something comes in |
| 09:05.24 | angryuser | schmidts, i have this for all of my peers, it is spamming so fast |
| 09:05.38 | angryuser | schmidts, like 5 pages a second |
| 09:06.14 | WIMPy | What kind of peers? |
| 09:06.52 | angryuser | WIMPy, sip, and provider |
| 09:06.53 | WIMPy | Do you experience any problems? |
| 09:07.03 | angryuser | WIMPy, no but it is pamming like hell |
| 09:07.52 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:08.02 | angryuser | http://pastebin.com/g3KwPLS7 5 pages a second of that WIMPy schmidts |
| 09:08.44 | WIMPy | Well sip debug seems the obvious idea. |
| 09:09.15 | WIMPy | Have you tried restarting? |
| 09:11.05 | angryuser | WIMPy, yes its the same, lets try sip debug, but i have to "script" and grep after as impossible to see realtime |
| 09:15.46 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 09:18.07 | angryuser | WIMPy, the errors i noticed presentign at least some interest http://pastebin.com/PsCpqvYV |
| 09:18.24 | angryuser | WIMPy, but there is nothing else really |
| 09:19.16 | angryuser | WIMPy, sip debug is fine |
| 09:28.25 | WIMPy | If it's not SIP, what else is happening? |
| 09:30.11 | schmidts | maybe a deadlock but this should go away after a restart |
| 09:51.34 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 10:02.23 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 10:07.53 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
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| 10:20.11 | *** part/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 10:24.05 | *** join/#asterisk AliRezaTaleghani (~shayne@unaffiliated/AliRezaTaleghani) |
| 10:24.21 | AliRezaTaleghani | hi all |
| 10:25.00 | AliRezaTaleghani | i need to do some special action, depend on the caller DTMF codes, and some queres on mysql |
| 10:25.10 | AliRezaTaleghani | i did some test by AGI |
| 10:25.28 | AliRezaTaleghani | but don't have any experience on AMI.. |
| 10:26.03 | AliRezaTaleghani | which way is fastes, easy and felaxible |
| 10:26.13 | AliRezaTaleghani | :-/ can somebody led me.... |
| 10:36.30 | *** join/#asterisk orly_owl (~DavoDinku@c122-108-89-112.sunsh1.vic.optusnet.com.au) |
| 10:36.40 | orly_owl | Price? http://www.digium.com/en/products/digital/te121.php |
| 10:45.51 | *** join/#asterisk hariom (~hariom@122.169.52.12) |
| 10:48.48 | *** part/#asterisk orly_owl (~DavoDinku@c122-108-89-112.sunsh1.vic.optusnet.com.au) |
| 10:48.59 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 10:50.30 | v1s | is there some kinda ata that allows u to hook up multible analog fones to an asterisk box? for like 10+ extentions? so dont have to use like sip fones can use standard phone |
| 10:53.20 | *** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 10:54.20 | *** part/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 10:57.45 | *** join/#asterisk eMBee (~eMBee@foresight/developer/pike/programmer) |
| 10:57.52 | eMBee | good evening |
| 10:58.51 | hariom | OFF TOPIC. Anybody looking to buy Asterisk Sangoma Card in India? I have a spare card (used only for 15 days for testing), with all original packaging. Also get free consultancy if you find problem setting it up. |
| 11:00.40 | eMBee | is trying to set up a new asterisk server using asterisknox and asteriskGUI. |
| 11:00.45 | eMBee | with one trunk line and a SPA942 sip phone. the phone is configured (works with old asterisk server) but on new one i get No matching peer found |
| 11:01.06 | *** join/#asterisk oej (~olle@91.183.59.105) |
| 11:02.03 | eMBee | asteriskgui shows the users, but in the cli 'sip show users' shows nothing |
| 11:03.12 | eMBee | not sure if that is relevant though |
| 11:05.11 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:06.38 | v1s | eMBee: what about: sip show peers |
| 11:07.12 | eMBee | v1s: 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 11:10.22 | v1s | did u add the user already ? |
| 11:10.36 | eMBee | yes |
| 11:10.51 | v1s | u reloaded the sip ? |
| 11:11.00 | v1s | sip reload |
| 11:11.07 | eMBee | reload in what way? |
| 11:11.22 | eMBee | restart asterisk itself? |
| 11:11.44 | v1s | u can do that or type in cli: sip reload |
| 11:12.48 | eMBee | oh! |
| 11:12.57 | eMBee | that changes something! |
| 11:13.05 | eMBee | thanks! |
| 11:13.11 | v1s | np |
| 11:15.21 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 11:18.36 | eMBee | now i get: Call from '123' to extension '12345678' rejected because extension not found in context 'DLPN_DialPlan' |
| 11:19.15 | eMBee | i set up the dial plan with _XXX. asuming that should allow all numbers with at least 3 or 4 digits |
| 11:20.22 | verywiseman | what is happened if i duplicate variable ,is it take last one or what? |
| 11:28.26 | v1s | eMBee: but I think u dialed more then 3 digits |
| 11:28.32 | v1s | u can try _X. |
| 11:28.46 | v1s | verywiseman: last one I belive |
| 11:29.00 | eMBee | v1s: yes, i dialed 8 digits |
| 11:31.55 | eMBee | using _X. now, still no dice |
| 11:32.48 | v1s | whats the message? |
| 11:33.00 | *** join/#asterisk ruyo (~psantos@a81-84-7-119.cpe.netcabo.pt) |
| 11:33.46 | eMBee | same, but i just restarted asterisk, and now i get a dialtone, hmmm |
| 11:34.18 | v1s | if u change ur dialplan |
| 11:34.19 | eMBee | the call does not get through |
| 11:34.28 | v1s | u have to in cli: dialplan reload |
| 11:34.36 | eMBee | aha |
| 11:34.37 | v1s | or restart asterisk |
| 11:34.39 | eMBee | ok |
| 11:37.32 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 11:37.39 | eMBee | so, now, i can dial, and after dialing the phone claims it is connected. what i hear is a continuous tone that sounds like the dialtone on an analog phone (so it would be the dialtone from the trunk i guess) |
| 11:38.06 | verywiseman | is it must [general] to be at the top of extensions.conf? |
| 11:39.12 | eMBee | does that mean that for some reason it connects to the trunk but does not actually send out the number i diealed? |
| 11:39.42 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:40.12 | eMBee | i don't see any more errors in the log other than nable to enable echo cancellation on channel 1 (No such device |
| 11:44.43 | eMBee | now, if i dial again on that dialtone, then i can actually make a call |
| 11:45.13 | eMBee | but why does ot not make the call directly? |
| 11:46.12 | v1s | eMBee: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 11:47.59 | eMBee | is reading |
| 11:49.03 | v1s | eMBee: theres lots of good info on that site ;) |
| 11:49.56 | eMBee | for a beginner like me there is to much info, hard to know what keywords to look for |
| 11:54.16 | v1s | yes |
| 11:54.21 | eMBee | hmm, naming a rule 'all' makes it hard to grep for in config files :-) |
| 11:58.26 | eMBee | hmm, maybe i set up the trunk wrong |
| 12:05.08 | eMBee | what do i reload to activate trunk changes? or is there a command to reload everything? |
| 12:06.03 | v1s | sip reload |
| 12:06.19 | eMBee | ah, ok, thanks |
| 12:12.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:13.46 | *** join/#asterisk rethus (~suther@p5087F23B.dip.t-dialin.net) |
| 12:14.05 | rethus | someone knows: can i phone with twinkle via capi (fritzubox-router? |
| 12:14.38 | [TK]D-Fender | rethus: * speaks CAPI. Twinkles is a SIP client/ So it sure sounds like "yes" to me. |
| 12:15.00 | rethus | i mean not over * i mean twinkle themself |
| 12:15.53 | rethus | someone told me ant-phone, but i have twinkle still installed... and if it would work, i didn't need to install ant-phone |
| 12:17.29 | [TK]D-Fender | rethus: CAPI isn't a VoIP protocol. So how is a SIP CLIENT going to speak it? |
| 12:19.22 | rethus | ok, thanks. i thought, maybe there is a plugin for twinkle i didn't know |
| 12:19.22 | [TK]D-Fender | rethus: As for rethus If that is a gateway as it it appears, it might work direct. GO TRY |
| 12:19.29 | eMBee | ok, my dialplan is exten = _X.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:0})},,trunk_1,), and trunkdial-failover-0.3 does s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1), so it is checking the length of my CID? |
| 12:20.05 | [TK]D-Fender | eMBee: That doesn't tell us enough to say "yes" |
| 12:20.36 | eMBee | hmm, ok |
| 12:20.49 | *** join/#asterisk bjornts (~Adium@it010225.klientdrift.uib.no) |
| 12:21.56 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 12:22.00 | [TK]D-Fender | eMBee: Axtually... it kinda does. Those are each priority 1. No variable gets copied anywhere, so no, it is not checking the callerid length anywhere |
| 12:22.50 | eMBee | what does the GotoIf do then? |
| 12:24.35 | [TK]D-Fender | eMBee: Exacty what it says. It tests some other double eval'd constant or dialplan var which we don't see set anywhere and jumps where it is told |
| 12:24.47 | [TK]D-Fender | eMBee: We ahve no idea what that thing holds |
| 12:24.54 | eMBee | the way i read this it means: if the length of FMCIDNUM is larger than 6, then it jumps to 1-fmsetcid |
| 12:25.22 | eMBee | yes, my question was specific on the jump condition, not what the jump results in |
| 12:26.02 | eMBee | 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) seems to set the caller id, which seems to make sense, if the id value is long enough, set it |
| 12:26.59 | eMBee | otherwise, 1, that means, next line? ie s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)? |
| 12:27.43 | eMBee | hmm, i don't have any callerid value on my trunk line |
| 12:27.54 | eMBee | maybe i should change that... |
| 12:27.56 | [TK]D-Fender | [08:25]<eMBee>1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) seems to set the caller id, which seems to make sense, if the id value is long enough, set it <- I don't see where this gets CALLED based on the 2 lines you gave us |
| 12:28.24 | [TK]D-Fender | eMBee: the one calling the macro didn't have that preceeding it, and the first line of the macro compares that value which isn't set anywhere. |
| 12:28.29 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 12:28.46 | [TK]D-Fender | eMBee: That other line of dialplan you just showed us doesn't seem to be referenced by either of those |
| 12:29.01 | eMBee | ax if the value is not set then the test should fail? |
| 12:29.03 | [TK]D-Fender | eMBee: May use some of the same values but isn't part of the execution |
| 12:29.39 | [TK]D-Fender | eMBee: Dunno.. it could be a CONSTANT defined earlier, or some global variable... but I don't see proof of either. |
| 12:30.33 | eMBee | ok, for now i am just trying to make sure i read the code right' |
| 12:30.34 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 12:31.13 | eMBee | so iff FMCIDNUM is set, its length is checked and it should go to 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) |
| 12:31.45 | eMBee | otherwise it should go to s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1) |
| 12:31.48 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 12:33.19 | [TK]D-Fender | eMBee: Looks like. |
| 12:33.38 | eMBee | thanks |
| 12:35.37 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 12:36.24 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 12:38.01 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:39.50 | eMBee | in _X.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:0})},,trunk_1,), ${ARG1} for trunkdial-failover-0.3 is ${trunk_1}/${,${EXTEN:0})}, is that correct? |
| 12:40.16 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 12:46.19 | [TK]D-Fender | eMBee: Yes, whatever that evals to |
| 12:46.56 | eMBee | ok |
| 12:48.29 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 12:48.48 | eMBee | that would be the next question, how can i find out what it ewvals too? i think at this point i should run the log in debug mode and watch |
| 12:49.06 | [TK]D-Fender | eMBee: You seem to be catching on... good... |
| 12:49.20 | eMBee | :-) |
| 12:49.59 | nextime | hello all. I have a strange problem after 2 days of internet link down where i have an asterisk server. It does register to a voip provider to get some incoming DID. There are 6 different register => in my sip.conf, with different credential buf the same host |
| 12:50.34 | nextime | now, two days ago all was working good. Today if i use only one register ( and others are commented ) |
| 12:50.51 | drmessano | Who is the provider? |
| 12:51.14 | nextime | all is working, but if i decomment more than 2 of them, registry goes in timeout and all peering ( also internal one ) get unreachable and it doesn't register anymore |
| 12:51.45 | nextime | drmessano : voip4biz, but it doesn't have problems as i have other 8 machines in different locations that are working good |
| 12:51.47 | *** join/#asterisk mintos (~mvaliyav@209.132.181.86) |
| 12:52.12 | [TK]D-Fender | nextime: Your peer has no relation to your register statements. You have connection issues period |
| 12:52.55 | nextime | [TK]D-Fender: i was thinking the same, but if i comment register it work good, if i decomment register it get to be unreachable |
| 12:53.12 | nextime | i've tryed many time and it is reproducible every time |
| 12:53.37 | [TK]D-Fender | nextime: Statement stands. |
| 12:54.11 | [TK]D-Fender | nextime: Qualify on the peer is its own thing (which you could flat-out DISABLE) |
| 12:54.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:54.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:54.50 | nextime | yes, but it is strange and unexpected |
| 12:55.04 | nextime | and anyway it doesn't solve the registry problem |
| 12:56.16 | [TK]D-Fender | nextime: Connection or provider issue. Take your pick |
| 12:56.38 | [TK]D-Fender | nextime: Maybe you could actually SHOW us your attempts. That would be novel.... |
| 12:56.52 | nextime | [TK]D-Fender : sadly i can exclude both, connection is working good as i can see with other protocols and tcpdump |
| 12:57.15 | nextime | and the provider is also good as i have other 8 * server registered and working |
| 12:57.28 | [TK]D-Fender | nextime: other protocols say nothing for a UDP protocol that gets massacred by packet-loss, etc. |
| 12:57.52 | [TK]D-Fender | nextime: Better show us some debug of the traffic you DO have on it |
| 12:58.06 | nextime | [TK]D-Fender : yes, but other protocols analyzed say if the network is working or not |
| 12:58.31 | nextime | anyway, i'm sure about network, the problem is somewhre on * i'm sure about that |
| 12:58.35 | nextime | also |
| 12:58.37 | [TK]D-Fender | nextime: Testing your windshield wipers doesn't mean the radio will work in your car. |
| 12:58.48 | [TK]D-Fender | nextime: Do NOT lump everything together. |
| 13:00.23 | hrhrhr | good afternoon, folks and welcome to #bizarre-analogies! |
| 13:00.56 | *** join/#asterisk ningia (~gain@109.69.131.226) |
| 13:01.05 | [TK]D-Fender | hrhrhr: ... putting the ANAL back into "analogy" |
| 13:01.15 | hrhrhr | :) |
| 13:02.45 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:03.30 | eMBee | is not getting anywhere |
| 13:04.34 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:05.32 | [TK]D-Fender | eMBee: In case nobody has told you yet, AsteriskGUI is broken, incomplete and unmaintained..... |
| 13:05.37 | ramindia | hello all |
| 13:05.53 | ramindia | can some one help me regarding TE121 card with asterisk.. the line not able to up |
| 13:05.58 | ramindia | i always get down status |
| 13:05.59 | eMBee | [TK]D-Fender: that is indeed new. |
| 13:06.06 | ramindia | light show green on the card |
| 13:06.15 | ramindia | Status: Provisioned, Down, Active |
| 13:06.22 | ramindia | any one here to help me on the same |
| 13:06.24 | [TK]D-Fender | eMBee: And its development stalled over a year ago. |
| 13:06.50 | ramindia | chan_dahdi.c:2790 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 13:06.52 | [TK]D-Fender | ramindia: What does your telco say? Is the line active on their side? |
| 13:07.29 | ramindia | they tested with bert meter i can see the call coming on and ringing |
| 13:07.46 | eMBee | oh, well, then i better switch |
| 13:08.03 | ramindia | when i connect to card i get busy tone |
| 13:08.36 | [TK]D-Fender | ramindia: Was this server working previously, or is this a new install? |
| 13:09.13 | ramindia | yes card was working, move to new location with new line |
| 13:09.51 | [TK]D-Fender | ramindia: I asked if it was working THERE. Sounds like a "no" |
| 13:09.53 | eMBee | is there an easy way to switch to freepbx and reset the config on the asterixnow cd? |
| 13:10.00 | eMBee | or is reinstall eaiser? |
| 13:10.06 | [TK]D-Fender | eMBee: Nope. Flush and replace |
| 13:10.26 | eMBee | [TK]D-Fender: you mean reinstall? ok |
| 13:10.31 | [TK]D-Fender | eMBee: What are your goals with this system BTW? |
| 13:10.43 | ramindia | yes |
| 13:11.03 | ramindia | it was working in old work palce.. |
| 13:11.14 | ramindia | move to new work place with new Line and new server installation |
| 13:11.22 | ramindia | i mean provider is different one |
| 13:11.36 | ramindia | i can provide if any logs required |
| 13:11.59 | eMBee | [TK]D-Fender: set up a new system from scratch, several trunk lines and a bunch of voip phones in the office |
| 13:12.15 | [TK]D-Fender | ramindia: Have you tested on any other hardware successfully there except for whatever this "bert" tester thing was? |
| 13:12.43 | [TK]D-Fender | eMBee: Planning on getting "creative" with its functionality? |
| 13:12.45 | ramindia | no we have only one card..not have option to test with any hardware.. |
| 13:13.00 | ramindia | the line going up and down.. |
| 13:14.01 | eMBee | [TK]D-Fender: not initially: the most creativity i'll get inot is making extensions work in callerID if that is even possible here, otherwise try to send different trunk numbers by callerid |
| 13:14.22 | [TK]D-Fender | ramindia: Pastebin your configs, "cat /proc/interrupts", and output of "dahdi_cfg -vvvv" |
| 13:14.27 | eMBee | since we need to assign different numbers to outgoing calls based on which phone makes the call |
| 13:14.48 | [TK]D-Fender | eMBee: that might be easy enough in the scope of their GUI |
| 13:15.04 | eMBee | i would hope so |
| 13:16.21 | eMBee | i know freepbx can set the callerid, but in that old box it ididn't work. however this box i inherited has a trixbox install that is ancient, so i didn't even try to debug, but went straight for reinstall, where i am now |
| 13:16.52 | ramindia | here is the output http://pastebin.com/ddTrBubZ |
| 13:17.24 | [TK]D-Fender | ramindia: 169: 1 1 2 4 1331381 9749 IO-APIC-level ohci_hcd:usb3, ohci_hcd:usb4, HDA Intel, wcte12xp0 <------ BAD. Your card should not be sharing an IRQ. Fix this |
| 13:18.24 | ramindia | can you suggest me the solution, how can i fix that |
| 13:18.55 | [TK]D-Fender | ramindia: Change slots, disable unneeded devices, etc |
| 13:19.07 | ramindia | in the bios you mean ? |
| 13:19.14 | [TK]D-Fender | ramindia: Yes |
| 13:19.22 | ramindia | ok will be right back after doing that |
| 13:19.53 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 13:21.28 | Katty | yawns |
| 13:21.32 | Katty | pamples things |
| 13:21.51 | [TK]D-Fender | Katty: pamplemouse? |
| 13:22.13 | [TK]D-Fender | missed an "s" ... dangit |
| 13:22.57 | Katty | [TK]D-Fender: http://www.tasteofhome.com/recipes/Chai-Tea-Latte |
| 13:23.06 | *** part/#asterisk rethus (~suther@p5087F23B.dip.t-dialin.net) |
| 13:23.59 | [TK]D-Fender | Katty: I call BS on that one. |
| 13:24.12 | [TK]D-Fender | Katty: That is way out... |
| 13:24.29 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 13:24.41 | [TK]D-Fender | Katty: Far better : http://www.2basnob.com/chai-tea.html |
| 13:24.43 | Katty | [TK]D-Fender: it's yummy. |
| 13:24.46 | Katty | oh? |
| 13:25.06 | Katty | too much work |
| 13:25.20 | [TK]D-Fender | Katty: note the 8 cardamom pods, 8 whole cloves, and the base of Darjeeling Tea |
| 13:25.45 | *** join/#asterisk Defraz (~Defraz@97-117-75-211.slkc.qwest.net) |
| 13:26.02 | Katty | [TK]D-Fender:yes, but it's too much work. i'd never make it |
| 13:26.14 | [TK]D-Fender | Katty: That's like saying baking a cake is too hard so you buy a store-made muffin and use that as the instructions for "making a cake" |
| 13:26.25 | Katty | exactly. |
| 13:26.41 | Katty | i will do what i want (= |
| 13:27.26 | [TK]D-Fender | Katty: You're starting to look Tragically White. Get your sorry ass out of the mid-west before it's too late! :p |
| 13:28.21 | drmessano | Yes, please head to the "west siiiide" or "east siiiide" as soon as possible |
| 13:28.47 | Katty | [TK]D-Fender: i just know i'll never make that recipe. |
| 13:28.58 | Katty | [TK]D-Fender: that's all there is too it |
| 13:30.12 | [TK]D-Fender | Katty: reminds me of when a Chinese friend of mine told me what a Tragically White friend of his said as a kid "We had Chinese food last night: macaroni & cheese with SOY SAUCE" |
| 13:30.20 | ramindia | [TK]D-Fender: hi, herre is the latest http://pastebin.com/cemgdKtz |
| 13:30.34 | ramindia | i do not see any conflict |
| 13:30.43 | [TK]D-Fender | ramindia: Certainly looks better. |
| 13:30.49 | [TK]D-Fender | ramindia: go look |
| 13:31.21 | ramindia | here it show down still http://pastebin.com/zpRJn3ev |
| 13:31.52 | drmessano | [TK]D-Fender, I had chinese food the other day, too. Sweet and Sour Chicken. Oh, wait. |
| 13:32.12 | [TK]D-Fender | ramindia: ask your telco to examine the line as you test and tell you what they see |
| 13:32.39 | [TK]D-Fender | drmessano: that is what the Chinese call "White-people food" |
| 13:33.22 | [TK]D-Fender | I also laugh at people who think General Tao Chicken is Chinese. Its all so sad... |
| 13:34.38 | drmessano | [TK]D-Fender, Yep. Make sure you serve it on the buffet next to the chocolate pudding and the puff pastry |
| 13:35.18 | coppice | [TK]D-Fender: sweet and sour chicken is perfectly valid cantonese food |
| 13:35.34 | drmessano | Not the way we make it here |
| 13:35.59 | ramindia | [TK]D-Fender: sure i will check with provider, here is the logs i see http://pastebin.com/hLw12RDR |
| 13:36.06 | [TK]D-Fender | drmessano: That would be a profiterole BTW |
| 13:36.15 | [TK]D-Fender | drmessano: Decidedly FRENCH |
| 13:36.47 | [TK]D-Fender | coppice: True, but looking at how much things change by the time most people here get to see it... |
| 13:37.10 | drmessano | Sugar paste + Orange flavoring = Orange Chicken, Sugar paste + Lemon flavoring = Lemon chicken, Sugar paste + Sweet and Sour flavoring = Sweet and Sour chicken. |
| 13:37.30 | drmessano | Just pull out the packet for what we're serving today |
| 13:37.40 | coppice | oh, *your* sweet and sour may be anything but cantonese.... though the US has a lot of places called Canton |
| 13:37.40 | [TK]D-Fender | ramindia: You may have MB issues.... |
| 13:38.08 | ramindia | mother board ? |
| 13:38.21 | [TK]D-Fender | coppice: that is also the brand of fondue base I buy :p |
| 13:38.28 | [TK]D-Fender | ramindia: Yes |
| 13:38.30 | drmessano | coppice: We put the "can't" in Cantonese |
| 13:39.49 | drmessano | Just like american pizza. Purely a product of teenage wet dreams and fast food era mass production |
| 13:40.26 | drmessano | It's no more "Pizza" than it is "toast with tomato jelly and cheese" |
| 13:40.59 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 13:41.11 | [TK]D-Fender | Most of the Cantonese food people see on this side of the planet is the Tragically White stuff unless you have a decent China-town... much prefer szechuan |
| 13:41.37 | [TK]D-Fender | drmessano: And in Domino's case with extra cardboard |
| 13:42.15 | drmessano | I am glad you said "with extra cardboard" and not "with cardboard" |
| 13:42.40 | FlashDeluxe | hi! can somebody help me please? I often get the error "Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)". I can call out and get calls in general, however, i get this error quite often :( |
| 13:42.41 | [TK]D-Fender | coppice: Fortunately I have a few really good Cantonese places here. |
| 13:43.06 | [TK]D-Fender | FlashDeluxe: Many telco's use that to mean "Busy" |
| 13:43.43 | *** join/#asterisk srini (~Srini@182.72.141.126) |
| 13:44.06 | srini | Hi room |
| 13:44.25 | FlashDeluxe | [TK]D-Fender But the line is clear?! |
| 13:44.31 | srini | I am a newbie trying to bring up a goautodial |
| 13:45.03 | [TK]D-Fender | FlashDeluxe: Referring to the TARGET line? Could also mean your telco is under-provisioned. |
| 13:45.21 | srini | While I try to agent login it says "Sorry, your phone login and password are not active in this system, please try again:" |
| 13:45.28 | srini | What could be the issue? |
| 13:45.30 | FlashDeluxe | [TK]D-Fender How can i find out if my telco causes these problems? :( |
| 13:45.58 | [TK]D-Fender | FlashDeluxe: Not sure, you could ask them to monitor the line as you place calls and see what they see on their end and to explain the nature of the given case... |
| 13:46.13 | [TK]D-Fender | srini: goautodial is NOT supported here |
| 13:46.33 | srini | [TK]D-Fender : Any other place? |
| 13:46.46 | [TK]D-Fender | srini: Go check their site and see what they offer |
| 13:47.09 | srini | [TK]D-Fender: Thanks! |
| 13:47.12 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
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| 13:50.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:26.14 | verywiseman | when i make conference room, i can make one extension for admin mode and another for other ,are that true? |
| 14:28.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:29.49 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:30.01 | [TK]D-Fender | verywiseman: You can make whatever extens you want |
| 14:30.12 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 14:32.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:33.18 | verywiseman | [TK]D-Fender, what is happened if there are 2 contexts have same name? |
| 14:33.46 | [TK]D-Fender | verywiseman: shouldn't happen, but I think they merge effectively |
| 14:33.51 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:34.08 | [TK]D-Fender | verywiseman: Go code your dialplan PROPERLY so this doesn't happen |
| 14:35.18 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:36.21 | *** join/#asterisk fireman_biff (~biff@65.48.132.153) |
| 14:38.43 | *** join/#asterisk BANSAL (~bansal@117.199.119.8) |
| 14:39.01 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-176-036.pools.arcor-ip.net) |
| 14:39.48 | verywiseman | [TK]D-Fender, is it must [general] to be at the top of extensions.conf? |
| 14:40.31 | [TK]D-Fender | verywiseman: Don't know if the location matters or not. Any reason you wouldn't put it there? |
| 14:40.33 | fireman_biff | hey guys is there a way to create a second trunk for a single PRI, for the purpose of limiting the number of calls going out on the trunk? |
| 14:42.15 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:42.19 | [TK]D-Fender | fireman_biff: You can group your channels any way you want for outgoing. Incoming control and ordering is up to the telco |
| 14:42.24 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:44.24 | fireman_biff | the thing is I don't want to group the channels (if I'm understanding that correctly). basically I have several PBXs, with one at the company's headquarters that needs to allow the other offices to make calls through it. I want to make sure that there's a limit on the amount of calls that can go out like that, but I want the channels available for the headquarters' use if nobody else is using them |
| 14:45.03 | fireman_biff | currently I limit the iax2 trunk going to HQ, but it would be better if I did the limit at HQ for all the offices combined |
| 14:45.59 | [TK]D-Fender | fireman_biff: then "core show function GROUP" would be it |
| 14:47.35 | fireman_biff | alright I'll look into it, thanks |
| 14:57.09 | ramindia | [TK]D-Fender: i have changed server, and inserted the card in to the other server, i still see same problem, any other suggestion |
| 14:57.22 | *** join/#asterisk BANSAL (~bansal@117.207.81.193) |
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| 15:06.16 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 15:06.17 | [TK]D-Fender | ramindia: Nope, I'm all out |
| 15:06.30 | ramindia | since the link is up and down |
| 15:06.40 | ramindia | the provider says he dont get any alarn |
| 15:06.48 | ramindia | since he is connected modem to modem |
| 15:06.53 | ramindia | he dont see that alarms |
| 15:07.28 | *** join/#asterisk bmoraca_work (~bmoraca@66.242.174.253) |
| 15:07.34 | Katty | hi bmoraca_work |
| 15:07.40 | bmoraca_work | hello |
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| 15:20.03 | *** join/#asterisk AndyML (~root@209.17.170.166) |
| 15:20.42 | AndyML | any Polycom users in the channel happen to remember how to get them to auto-answer directly to the headset, instead of the speaker phone? |
| 15:21.19 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 15:21.44 | pif | hi, what is the best 4 port ISDN card, digium or beronet? |
| 15:22.09 | Katty | pif: i've never heard of beronet. |
| 15:23.00 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 15:24.41 | WIMPy | pif: I'd go for something that also works with the kernel drivers like Junghanns. |
| 15:25.31 | pif | you mean misdn? |
| 15:25.43 | WIMPy | Or is beronet DDC based as well? |
| 15:25.53 | WIMPy | misdn2 to be precise, yes. |
| 15:26.07 | WIMPy | CCD |
| 15:26.42 | pif | misdn2 is not in kernel yet, right? |
| 15:26.49 | pif | it's misdn1 |
| 15:26.52 | AndyML | for the record, "Headset Memory Mode" allows inbound calls to auto-answer to the headset. thanks! |
| 15:27.14 | WIMPy | misdn2 has been in the standard kernel for two years I think. |
| 15:27.24 | WIMPy | misdn1 was abandoned. |
| 15:27.39 | WIMPy | gotta go |
| 15:30.36 | pif | oki doki |
| 15:31.38 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 15:37.46 | pif | but chan_misdn is still misdn1 (it seems) |
| 15:38.04 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
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| 15:53.58 | ruyo | pif: Yes, chan_misdn is for mISDNv1. |
| 15:54.53 | ruyo | mISDNv2 is in the standard kernel as of 2.6.27, which is after CentOS and Debian's stable versions. |
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| 16:02.55 | *** join/#asterisk oryxtec (~test@119.152.226.58) |
| 16:03.24 | oryxtec | hi all one small quick question ... which protocol is best for voice is't sip or iax? |
| 16:03.27 | oryxtec | which one is good |
| 16:03.33 | bmoraca_work | yes. |
| 16:03.50 | [TK]D-Fender | oryxtec: IAx is good when you NEED it, otherwise jsut stick with SIP |
| 16:04.42 | oryxtec | humm |
| 16:04.48 | leifmadsen | ~bestquestions |
| 16:04.48 | infobot | it has been said that bestquestions is "see ~thebestquestions" |
| 16:04.53 | leifmadsen | ~thebestquestions |
| 16:04.53 | infobot | i heard thebestquestions is Whenever you ask a "what is the best..." or "who is the best..." type questions, you're asking for trouble, and possibly may be called a troll. These types of questions do not have answers. Your best bet is to rephrase the question as, "What kind of experience do people have with..." or "Who has experience with...". |
| 16:04.53 | oryxtec | when usually do we need IAX? |
| 16:06.24 | bmoraca_work | oryxtec: when you have a fair number of calls trunked between asterisk systems that will generally not need to go outside they system of asterisk systems, because it has less overhead than the equivalent number of SIP channels...or if you have trouble making SIP work over NAT. |
| 16:07.01 | leifmadsen | I always use SIP in pretty much all situations |
| 16:07.14 | leifmadsen | bmoraca_work suggestions of when to use IAX2 are accurate though |
| 16:07.52 | bmoraca_work | i've never had a need to use IAX. SIP works just fine |
| 16:08.07 | leifmadsen | same |
| 16:08.28 | leifmadsen | in fact I've taken it out in situations where IAX was not working, and then switching to SIP solved the customers problems |
| 16:08.53 | *** join/#asterisk bongfrog (~quassel@2001:470:f1bc:1:213:2ff:fe67:39c5) |
| 16:08.57 | bmoraca_work | the benefit to SIP is that it's (mostly) widely supported. so, I can configure my asterisk system once and use almost any piece of equipment |
| 16:09.10 | oryxtec | humm |
| 16:09.18 | leifmadsen | hums as well |
| 16:10.27 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 16:10.49 | oryxtec | i don't understand one thing... why skype voice quality is soo soo good.. i have hosted server in UK if i registed 2 100 and 200 extensions on my asterisk box and when i dial that ext 200.. voice quality is not really good.. but when i use skype.. on system computers and same internet speed voice is very good |
| 16:11.00 | oryxtec | plz could any one tell me why is that |
| 16:11.38 | bmoraca_work | where are you located? generally, you'd be looking at a latency issue (>150ms will cause quality issues) |
| 16:11.53 | oryxtec | i m in pakistan |
| 16:12.01 | bmoraca_work | well, there's your problem |
| 16:12.03 | bmoraca_work | you're too far away |
| 16:12.12 | oryxtec | but |
| 16:12.14 | bmoraca_work | latency is too great to your server |
| 16:12.27 | oryxtec | skype latency is also >150 |
| 16:12.36 | oryxtec | but their voice is too dam good |
| 16:12.37 | oryxtec | :'( |
| 16:12.55 | bmoraca_work | skype doesn't use SIP. skype is likely a buffered protocol. SIP is real-time UDP. |
| 16:13.29 | oryxtec | soo they don't use SIP? then wht exact protocl do they use? |
| 16:13.33 | bmoraca_work | if skype uses TCP, it's not possible for packets to arrive out-of-order, which is what latency and jitter do, and cause audio quality issues |
| 16:13.40 | Chainsaw | oryxtec: Nobodoy knows, it's quite heavily encrypted. |
| 16:13.48 | bmoraca_work | from what I understand, they use a proprietary protocol |
| 16:13.49 | oryxtec | humm |
| 16:14.00 | Chainsaw | oryxtec: Nobody, even. One day I'll learn to type. |
| 16:14.39 | Chainsaw | oryxtec: Personally I find Skype quite sinister. They charge for all infrastructure and it is not clear who listens in. |
| 16:14.45 | oryxtec | please recomend me wht should i do to make my voice quality good |
| 16:15.00 | carrar | QoS |
| 16:15.12 | Chainsaw | oryxtec: You need a link with low *and consistent* latency. |
| 16:15.21 | Chainsaw | oryxtec: That may take hosting the server closer to where you and your customers are. |
| 16:15.42 | oryxtec | right .. |
| 16:16.25 | oryxtec | i mean to say.. can i change my server settings.. may be use IAX with g729 rather then SIP may be it will help to increase my voice quality |
| 16:16.27 | oryxtec | ? |
| 16:17.41 | oryxtec | any ideas? please? |
| 16:18.53 | drmessano | IAX will not help you |
| 16:18.59 | drmessano | G729 is covering the problem |
| 16:19.43 | oryxtec | humm |
| 16:19.58 | oryxtec | soo you recomend me sip with g729 |
| 16:21.46 | oryxtec | one last thing i m planning to buy this server http://www.serverloft.com/dedizierte-server/server-details.php?products=2 |
| 16:22.01 | oryxtec | please tell me using g729 how many calls this server can support |
| 16:22.01 | oryxtec | ? |
| 16:24.26 | carrar | ~book |
| 16:24.26 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:25.41 | *** join/#asterisk wierdo (jimmy@77.78.3.197) |
| 16:25.58 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:26.21 | bongfrog | good morning all: I have a new te410p that I am putting in and existing asterisk server but keep getting 'Unknown device 1410 (rev 02)" when lspci.. zap 1.4.12.9. states that the card is supported. Any Idea where I should look? |
| 16:26.48 | bongfrog | I have tried a update-pciids with no change |
| 16:28.00 | Chainsaw | bongfrog: As long as you see a driver associated with the device, the IDs being unknown won't hurt. |
| 16:28.43 | Chainsaw | oryxtec: Are you planning to deploy in Germany or in the US? |
| 16:30.50 | bongfrog | Chainsaw: I have the wct4xxp module insereted and zaptel no see em.... |
| 16:33.13 | Chainsaw | Okay then. |
| 16:40.17 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 16:45.57 | *** join/#asterisk m_tadeu (~quassel@89-181-59-221.net.novis.pt) |
| 16:50.38 | Nugget | http://macnugget.org/photos/2007gt3rs/bofh_rtfm <-- from this weekend (BOFH and RTFM) |
| 16:51.30 | drmessano | Sweet, but I call "Photoshop" on it |
| 16:51.40 | drmessano | No BOFH would use an Apple anything |
| 16:51.55 | Nugget | it's my car, dude. :) |
| 16:52.26 | leifmadsen | steals Nugget's car |
| 16:52.28 | drmessano | I guess it looks like I am calling you a fraud then, doesn't it? |
| 16:52.39 | Nugget | guess so :( |
| 16:52.40 | leifmadsen | I should get the RTFM plate :) |
| 16:53.05 | drmessano | Either way, that's pretty funny. I must tweet it. |
| 16:55.33 | p3nguin_ | Your car is an Apple? |
| 16:56.11 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 16:57.18 | [TK]D-Fender | p3nguin_: No, but possibly a lemon :p |
| 16:58.38 | *** join/#asterisk ritzt3ch (~ritztech@ip65-47-226-86.z226-47-65.customer.algx.net) |
| 16:59.13 | ritzt3ch | Soo TK i just switched to sip trunk inbound and it work instant .... (has to be something in DAHDI timing ? ) |
| 16:59.53 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 17:02.08 | [TK]D-Fender | ritzt3ch: ... I don't know what I'm comparing here, and you are telling me less with each time you bring up your various projects... |
| 17:02.30 | m_tadeu | ok, I read lots documents about sip+nat, followed them and still have a problem. Can't hear any sound when a user goes into a queue and an agent answers. What can I do to check what is going wrong? |
| 17:02.33 | ritzt3ch | haha sorry my mind thinks like everyone knows what the hell im talking about.... |
| 17:02.47 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:03.35 | [TK]D-Fender | m_tadeu: Describe your environment |
| 17:04.42 | m_tadeu | astrisk server and agents behind nat and clients outside. all using sip softphones |
| 17:06.24 | [TK]D-Fender | m_tadeu: Forward 5060,10000-20000 all UDP to your server, ensure you have NAT=YES, EXTERNIP=1.2.3.4 (your IP), and CANERINVITE=NO under [general]. |
| 17:06.33 | [TK]D-Fender | canreinvite* |
| 17:06.54 | m_tadeu | ah if the client calls an agent disrectly to it's extension, all goes just fine |
| 17:07.18 | ritzt3ch | haha i had Dahdi FXS inbound and it took 3 seconds After i dialed the 4 digits To even start processing the Dial plan |
| 17:07.46 | bmoraca_work | ritzt3ch: that's the nature of analog |
| 17:07.53 | ritzt3ch | Couldnt find it so we reverted to Sip trunks instead .... dam shit worked instant |
| 17:07.56 | m_tadeu | [TK]D-Fender: everything is done like that...only diff is I have a name in externip |
| 17:08.14 | [TK]D-Fender | m_tadeu: Not legal. That is what externhost is for |
| 17:08.33 | [TK]D-Fender | m_tadeu: also you'd need "externrefresh" set to some interval to keep it updated |
| 17:08.41 | leifmadsen | Nugget: damn you! I wish my xbox was working so I could play some forza :) |
| 17:08.56 | m_tadeu | [TK]D-Fender: I had both...wouldn't it work? |
| 17:08.58 | [TK]D-Fender | ritzt3ch: I've have to see the full comparison. |
| 17:09.05 | [TK]D-Fender | m_tadeu: Don't do both |
| 17:09.17 | [TK]D-Fender | m_tadeu: When you're done, PASTEBIN it so we can see. |
| 17:09.28 | m_tadeu | [TK]D-Fender: got it...thx |
| 17:09.51 | ritzt3ch | Yea after going through so many steps i had 2 conclusions WHY ... 1 CID (which i read carrier takes a second to send the CID and the NAME) and Asterisk couldnt find it. OR chan_dahdi.c matchdigittimeout=3000 (but didnt knwo which one) .... |
| 17:10.20 | [TK]D-Fender | ritzt3ch: CID != DTMF. Sounds like you used a bad pattern |
| 17:11.09 | Nugget | leifmadsen: heh |
| 17:13.25 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 17:14.10 | m_tadeu | [TK]D-Fender: I'm also setting qualify=yes, is this correct? |
| 17:15.22 | [TK]D-Fender | m_tadeu: for your phones, yes |
| 17:19.07 | Katty | mmm, ham and swiss sammich |
| 17:19.21 | p3nguin_ | hot or cold? |
| 17:19.25 | Katty | cold. |
| 17:19.25 | [TK]D-Fender | yes |
| 17:19.27 | Katty | i'm lazy. |
| 17:19.43 | p3nguin_ | I'd rather have it hot and on a toasted bun. |
| 17:19.56 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 17:19.59 | cusco | hi... |
| 17:20.12 | Katty | ohai |
| 17:20.15 | cusco | queue members have device being Local/xxx |
| 17:20.32 | p3nguin_ | Orly? |
| 17:20.33 | cusco | so it performs some dialplan before dialing sip/xxx |
| 17:20.35 | cusco | ok... |
| 17:20.36 | cusco | so |
| 17:21.16 | *** join/#asterisk Bartockbatz (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 17:21.18 | cusco | what is the common'est way for queues to identify that the peer is in a call and do not dial him |
| 17:21.22 | cusco | ? |
| 17:21.49 | cusco | we were using group count |
| 17:22.10 | cusco | then my boss tells me that we should use call-limit |
| 17:22.30 | cusco | but... shouldn't the queue know that the devie is in use or soemthing? |
| 17:22.37 | cusco | device |
| 17:23.02 | cusco | :/ |
| 17:23.37 | [TK]D-Fender | cusco: You would have to specify the state device to use in AddQueueMember. That is the only way |
| 17:23.52 | *** join/#asterisk Tim_Toady (~moi@77.49.105.80.dsl.dyn.forthnet.gr) |
| 17:24.30 | cusco | the members are realtime... so is that a flag in the database? |
| 17:24.50 | [TK]D-Fender | cusco: dunno.... |
| 17:25.32 | cusco | so in AddQueueMember if I say the device state is "in use" queue won't dial to him, right? |
| 17:26.01 | cusco | but then when the call is hung up, how does it become "not in use" again? |
| 17:26.17 | AliRezaTaleghani | ChannelZ: hi, do u have time for me? |
| 17:26.22 | *** join/#asterisk [T]ank (~ckwall@c-71-195-199-101.hsd1.ut.comcast.net) |
| 17:27.01 | AliRezaTaleghani | i have a problem with the Macro argument of the Queue Application |
| 17:27.43 | p3nguin_ | cusco: I think you need to specify the actual device as the state interface. |
| 17:27.59 | AliRezaTaleghani | as it's mentioned on it's documents, Macro should be run on the Caller channel |
| 17:28.16 | AliRezaTaleghani | not the member of the Queue (Agent of that Queue) |
| 17:28.18 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 17:28.53 | AliRezaTaleghani | but as i use it's running on the Member channel |
| 17:28.56 | AliRezaTaleghani | :( |
| 17:29.06 | [TK]D-Fender | AliRezaTaleghani: Show us your docs,your configs, and your attempt |
| 17:30.30 | AliRezaTaleghani | [TK]D-Fender: i had posted something about it on Elastix forum |
| 17:30.34 | AliRezaTaleghani | http://www.elastix.org/en/component/kunena/19-call-center/61648-playback-agent-name--id-to-the-caller.html |
| 17:31.25 | AliRezaTaleghani | is it enough? |
| 17:32.38 | p3nguin_ | cusco: I don't actually use AddQueueMember(), but it looks like you would do something like AddQueueMember(queue-name,Local/123@queue-devices,,,,SIP/george-phone) to dial Local/123@queue-devices as the queue member and check the state of SIP/george-phone. |
| 17:33.26 | cusco | p3nguin_: I don't use addqueuemember neither. We use realtime mysql, and we insert it into mysql.... |
| 17:34.36 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 17:35.15 | [TK]D-Fender | AliRezaTaleghani: Show me where it says that it calls the macro for the calling channel |
| 17:35.28 | [TK]D-Fender | cusco: Too bad. |
| 17:35.46 | AliRezaTaleghani | [TK]D-Fender: ok, in a second |
| 17:36.47 | cusco | so... there is no way to specify the state...? |
| 17:36.59 | cusco | how do people using realtime configuration do? |
| 17:37.14 | [TK]D-Fender | cusco: My gues is "The don't" |
| 17:37.18 | [TK]D-Fender | they* |
| 17:38.55 | generalhan | hey all ... i am having some issues getting a BLF line to work on an Aastra 480i. the BLF line appears on the softkey, but does not update when the line is active. i was wondering if anyone had a second to look at my config to see if i did anything totally wrong, though i have made this work in a similar fashion for other lines before. http://pastebin.com/2bi8YVBx |
| 17:50.49 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:53.08 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 17:55.14 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 17:56.28 | *** part/#asterisk fireman_biff (~biff@65.48.132.153) |
| 17:56.47 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 17:57.12 | Bartockbatz | hey - anyone see this before: |
| 17:57.57 | p3nguin_ | waits impatiently |
| 17:58.35 | Naikrovek | is going to slaughter his brother |
| 17:58.37 | Naikrovek | grrr. |
| 17:58.48 | Bartockbatz | it's coming |
| 17:58.58 | p3nguin_ | That's what she said. |
| 17:59.02 | Naikrovek | he got me banned from Steam (and the beloved TF2) |
| 17:59.05 | citywok | lol |
| 18:00.51 | *** join/#asterisk clintc (~clintc@n128-227-99-133.xlate.ufl.edu) |
| 18:00.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:02.07 | Bartockbatz | http://asterisk.pastebin.com/iqmk0fuZ |
| 18:02.37 | Bartockbatz | any suggestions would be greatly appreciated! |
| 18:02.40 | p3nguin_ | Seems pretty standard. |
| 18:02.58 | p3nguin_ | Fix the sip definition. |
| 18:03.40 | Bartockbatz | okay - next dumb question - can you point me on documentation on SIP definition please |
| 18:03.52 | Naikrovek | sip RFC, whatever number that is |
| 18:04.26 | p3nguin_ | There's plenty of useful info in the sample sip.conf and on voip-info.org about how to configure a sip peer for your phone. |
| 18:04.34 | AliRezaTaleghani | :-/ where is [TK]D-Fender |
| 18:04.46 | Bartockbatz | okay - I shall look there |
| 18:07.43 | Katty | hmmmmm. naptime me thinks |
| 18:10.26 | *** join/#asterisk squidly (~squidly@HoodLUG/member/squidly) |
| 18:12.05 | *** join/#asterisk mxmasster (~mxmasster@207.171.12.197) |
| 18:15.42 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 18:16.36 | citywok | I'm having issues with orphaned calls that last 100,000+ seconds before i go and hang them up. any ideas? |
| 18:17.05 | [TK]D-Fender | aliwhere is my PASTEBIN? |
| 18:17.12 | [TK]D-Fender | dangir |
| 18:22.24 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 18:22.45 | *** join/#asterisk fofware (~Fabian@186.124.144.194) |
| 18:29.48 | leifmadsen | citywok: look at session-timers and rtp timers in sip.conf.sample |
| 18:36.14 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 18:36.21 | [sr] | hi WIMPy, andread answered me :) |
| 18:37.47 | cusco | hmm for some reason group() or group_count() is not working... |
| 18:38.43 | cusco | dialplan has: Set(GROUP()=${EXTEN}); |
| 18:39.24 | [TK]D-Fender | cusco: PASTEBIN |
| 18:39.26 | cusco | ok hold |
| 18:42.10 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 18:42.41 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:43.43 | m_tadeu | [TK]D-Fender: I did the testing. One thing I noticed now is when I "sip show peers", the one outside had NAT set to N...the other peers don't have anything |
| 18:44.03 | [TK]D-Fender | m_tadeu: Not relevant |
| 18:44.31 | [TK]D-Fender | m_tadeu: for this part... now show some actual failed calls/reg, etc |
| 18:45.30 | m_tadeu | [TK]D-Fender: nothing shows up in the logs nor console..where else can I check that? |
| 18:45.35 | p3nguin_ | m_tadeu: The "N" means you have nat=yes set for the peer. |
| 18:46.01 | m_tadeu | p3nguin_: ok...thx :) |
| 18:46.36 | cusco | [TK]D-Fender: http://paste.debian.net/93252/ |
| 18:47.00 | cusco | but group_count returns always 1 |
| 18:47.06 | cusco | even if the peer is taken |
| 18:47.42 | [TK]D-Fender | cusco: AEL garbage. show me it fail in USE |
| 18:47.47 | cusco | ok |
| 18:50.00 | cusco | http://paste.debian.net/93254/ here - core show channels tells me that peer 633 is in a call |
| 18:50.34 | cusco | and it still rang |
| 18:51.03 | cusco | - Executing [633@agents:8] NoOp("SIP/150-00002be7", "633 is NOT busy. GROUP_COUNT=1") in new stack |
| 18:52.21 | tzanger | hmm, using a spa2102 instead of the old tdm400 is mostly win |
| 18:54.19 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-14.emserv.ru) |
| 18:54.39 | cusco | core show channel SIP/bla... shows Call Group: 0 |
| 18:54.42 | cusco | not sure if that is it |
| 18:55.00 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-14.emserv.ru) |
| 18:55.42 | [TK]D-Fender | cusco: And where do I see that this is wrong? |
| 18:55.52 | [TK]D-Fender | [14:51]<cusco>- Executing [633@agents:8] NoOp("SIP/150-00002be7", "633 is NOT busy. GROUP_COUNT=1") in new stack <- what makes this wrong? |
| 18:57.31 | cusco | he is already in a call, so that ran before. Now should return 2 |
| 18:57.55 | [TK]D-Fender | cusco: I see nothing that proves anything |
| 18:58.05 | cusco | ... |
| 18:58.13 | cusco | how can I show you that he is already in a call? |
| 18:58.18 | cusco | o let me wait for someone to get a call |
| 19:00.34 | cusco | got iy |
| 19:00.36 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 19:01.23 | cusco | [TK]D-Fender: http://paste.debian.net/93258/ |
| 19:01.45 | cusco | line 9 and 93 on pastebin |
| 19:01.58 | cusco | - Executing [608@agents:6] NoOp("Local/608@agents-67c7;2", "GROUP_COUNT of 608: 1") in new stack |
| 19:02.05 | cusco | - Executing [608@agents:6] NoOp("SIP/150-00002c2a", "GROUP_COUNT of 608: 1") in new stack |
| 19:02.31 | bmoraca_work | the call had already hung up before your second call had been placed |
| 19:02.44 | bmoraca_work | lines 60-68 |
| 19:02.49 | cusco | oops |
| 19:03.05 | [TK]D-Fender | reaches for his rusty-nail upgraded ClueBat (tm) |
| 19:03.23 | cusco | :( |
| 19:03.35 | cusco | ok Ill get another one.. let me just find out why he answered and hang up |
| 19:04.49 | cusco | ?! |
| 19:04.57 | cusco | he is still with the caller |
| 19:07.54 | m_tadeu | [TK]D-Fender: http://paste.debian.net/93260/ |
| 19:08.23 | cusco | why the hell is the call hanging up.. and he is still with me... ??? |
| 19:08.46 | cusco | perhaps that channel is hung but he stays with the briged channel? |
| 19:08.52 | [TK]D-Fender | No |
| 19:09.00 | [TK]D-Fender | cusco: You are not looking at the right things clearly |
| 19:09.12 | cusco | Im sure Im not... |
| 19:09.43 | cusco | I don't understand why is it going to h@agents |
| 19:10.45 | [TK]D-Fender | m_tadeu: and what is that supposed to tell me? |
| 19:10.55 | cusco | should I use group(cat)=bla ? |
| 19:11.39 | m_tadeu | [TK]D-Fender: that in the logs everything looks fine...at least to me |
| 19:12.16 | [TK]D-Fender | cusco: that isn't going to save you |
| 19:12.59 | cusco | what should I be looking at? |
| 19:13.16 | cusco | should I set OUTBOUND_GROUP ? |
| 19:14.58 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 19:16.04 | zoid_ | Hi, I'm new to asterisk and setting up my very first IVR, I have a question: If I use the command WaitExt, the user can interrupt the menu dialing the desired extension |
| 19:16.12 | zoid_ | however, I can't change the context |
| 19:16.35 | zoid_ | so, all my extensions, including those in submenus have to be in the same context |
| 19:16.37 | zoid_ | is that right? |
| 19:16.55 | [TK]D-Fender | zoid_: It will look in the context your IVR is in. |
| 19:17.17 | [TK]D-Fender | zoid_: You can however INCLUDE other contexts in there to avoid the duplication you are probably thinking you had to do |
| 19:17.17 | cusco | zoid_: you can use goto contezxt,exten,prioriti |
| 19:17.24 | p3nguin_ | zoid_: You want BackGround() for that. WaitExt() sits and waits silently for input for the specified time. |
| 19:17.28 | [TK]D-Fender | zoid_: And NO, you do NOT want Goto() |
| 19:17.43 | [TK]D-Fender | p3nguin_: CAFFEINE. Get some :p |
| 19:18.00 | zoid_ | the other option was to use playback, read, gotoif |
| 19:18.13 | p3nguin_ | Playback() does not allow interruption from the caller. |
| 19:18.17 | zoid_ | but with that setup the user can't interrupt the manu, which is annoing |
| 19:18.21 | zoid_ | exactly |
| 19:18.28 | p3nguin_ | BackGround() does! |
| 19:18.59 | zoid_ | p3nguin_: but it jumps to extensions in the same context, how do I reuse the extensions 1,2,3 in the sub menus? |
| 19:19.12 | zoid_ | do you understand my question? |
| 19:19.38 | p3nguin_ | I understand it... but do you? |
| 19:19.40 | [TK]D-Fender | zoid_: Your sub-menus should be IVR's in SEPARATE contexts |
| 19:19.52 | zoid_ | p3nguin_: not sure :P |
| 19:20.05 | p3nguin_ | Each context of the IVR should have the extensions 1, 2, and 3 (if you have three choices). |
| 19:20.39 | zoid_ | yes, but how do I make it jump from one context to another while I allow the user to interrupt the menus |
| 19:20.42 | p3nguin_ | If you need 1, 2, and 3 more than one time in the IVR, you'll create another context or it won't work. |
| 19:21.05 | p3nguin_ | And you'll use BackGround() to allow the interruption. |
| 19:21.56 | zoid_ | I'm sorry I'm not understanding this, but background does not know of other contexts |
| 19:22.27 | p3nguin_ | Nothing knows about other contexts until you create extensions that take you to them. |
| 19:22.51 | [TK]D-Fender | zoid_: What are you not following here? |
| 19:22.55 | p3nguin_ | exten => 1,1,Goto(context1,s,1) |
| 19:22.58 | zoid_ | I can jump to another context using Goto and GotoIf, that works |
| 19:23.07 | p3nguin_ | exten => 2,1,Goto(context2,s,1) |
| 19:23.14 | p3nguin_ | (just as an example) |
| 19:23.14 | zoid_ | yes, I understand that |
| 19:24.18 | p3nguin_ | Then what's the problem? |
| 19:24.18 | zoid_ | but I want to use Background, not Goto, to allow interruption, but Background needs that the extension the user dials is defined in the same context |
| 19:24.30 | p3nguin_ | BackGroun() plays sound files. |
| 19:24.33 | [TK]D-Fender | zoid_: I already answerd you on this |
| 19:24.37 | zoid_ | (also, please excuse my english) |
| 19:24.40 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 19:24.41 | p3nguin_ | it does not take you to another context. |
| 19:24.53 | [TK]D-Fender | [15:17]<[TK]D-Fender>zoid_: You can however INCLUDE other contexts in there to avoid the duplication you are probably thinking you had to do |
| 19:25.08 | cusco | ok that worked |
| 19:25.15 | cusco | outboud_group |
| 19:25.15 | p3nguin_ | And I even answered it in another way. |
| 19:25.26 | p3nguin_ | So now you have two possibilities. |
| 19:25.27 | nny | quick polycomm question, is the msg.mwi.1.subscribe="" required to have the PBX ip for proper VM notifications on a polycomm? troubleshooting intermittent mix ups between the phone MWI state and the PBX |
| 19:25.34 | zoid_ | [TK]D-Fender: but if I include 2 contexts with the same extension wouldn;t it conflict? |
| 19:25.55 | p3nguin_ | Contexts keep extensions separate. |
| 19:26.00 | [TK]D-Fender | zoid_: Yes, they would. Now why WOULD you do something ilke that? |
| 19:26.37 | p3nguin_ | They remain separate until until you create an association with another context using include, or a goto, etc. |
| 19:26.39 | [TK]D-Fender | zoid_: contexts are searched in INCLUDE order. so you might not get the one you're hoping for based on the order |
| 19:27.03 | zoid_ | Ok, I understand that |
| 19:27.11 | p3nguin_ | I doubt you. |
| 19:27.19 | zoid_ | p3nguin_: ;) |
| 19:27.26 | nny | ahh nm |
| 19:27.57 | [TK]D-Fender | nny: And that would be "no" |
| 19:28.42 | nny | [TK]D-Fender: yeah heh, figured it out. Context mismatch. (it's freepbx...) |
| 19:28.42 | p3nguin_ | zoid_: The short answer is DO NOT INCLUDE MULTIPLE CONTEXTS WITH DUPLICATE EXTENSIONS. |
| 19:29.11 | p3nguin_ | The long answer was already provided over the past ten minutes. |
| 19:29.15 | nny | [TK]D-Fender: has mailbox=XXXX@device in sip.conf, but "default" as the mailbox context. Just a configuration error on the installer's part |
| 19:29.20 | zoid_ | p3nguin_: ok |
| 19:29.34 | nny | [TK]D-Fender: apart from using freepbx :D |
| 19:29.41 | p3nguin_ | If you have additional questions, pastebin your dialplan so we can tell you why it doesn't work. |
| 19:29.54 | zoid_ | p3nguin_: right away, thanks |
| 19:31.40 | nny | [TK]D-Fender: gah.. grrr. Nm, freepbx symlinks device to default. .. ok off to #freepbx. Just wondering if it was a polycomm config issue |
| 19:32.14 | [TK]D-Fender | nny: Symlinks are for the files.. if the context names don't match it doesn't matter if the files follow |
| 19:32.22 | [TK]D-Fender | nny: It won't look for them |
| 19:32.28 | nny | [TK]D-Fender: yeah got ya |
| 19:32.39 | nny | [TK]D-Fender: will change it. thanks |
| 19:32.40 | [TK]D-Fender | nny: Go shoot them now. |
| 19:32.49 | nny | [TK]D-Fender: will do. Any particular size of ammo? |
| 19:33.06 | [TK]D-Fender | nny: .50 BMG should do |
| 19:35.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:35.35 | nny | [TK]D-Fender: installer error. found it |
| 19:36.16 | m_tadeu | ok, I ran out of ideas on how to check what the problem is...stil no sound goind in/out when an agent answers to a user from a queue |
| 19:36.38 | zoid_ | ok, here is my dialplan, without using Background http://pastebin.com/ATeuTATW |
| 19:37.17 | zoid_ | It works, now I want to use Background instead of MP3Player/Playback |
| 19:37.35 | [TK]D-Fender | zoid_: Read is NOT an IVR <- |
| 19:37.41 | [TK]D-Fender | zoid_: WAITEXTEN <- |
| 19:38.09 | [TK]D-Fender | zoid_: So Background() your prompts and call WaitExten() |
| 19:38.56 | zoid_ | [TK]D-Fender: In the first case, to replace line 11, how do I do it with Background and WaitExten? |
| 19:39.01 | zoid_ | sorry |
| 19:39.02 | zoid_ | not 11 |
| 19:39.16 | zoid_ | 12 |
| 19:39.43 | [TK]D-Fender | zoid_: exten => 1,1,Goto(argentina,s,1) |
| 19:39.53 | [TK]D-Fender | zoid_: for your options, and ditch all the READ's |
| 19:40.01 | *** join/#asterisk windback (~quassel@200.123.180.65) |
| 19:40.04 | [TK]D-Fender | zoid_: And do the Bacground and Waitexten like I told you to |
| 19:40.14 | zoid_ | [TK]D-Fender: I understand now |
| 19:40.22 | nny | another polycomm question. any reason why the phone call volume resets after each call? |
| 19:40.32 | zoid_ | thank you very much for you patience |
| 19:40.38 | [TK]D-Fender | nny: In your sip.cfg look up "persist" |
| 19:40.46 | [TK]D-Fender | nny: You can set each independently |
| 19:40.54 | nny | [TK]D-Fender: thanks |
| 19:42.09 | windback | One question: If I have a sip user called, for example gyl586 calling over DAHDI channel, I have the following message in CLI: WARNING[23385] chan_dahdi.c: Unrecognized prilocaldialplan NPI modifier: g |
| 19:42.13 | windback | WARNING[23385] chan_dahdi.c: Unrecognized prilocaldialplan NPI modifier: y |
| 19:42.17 | windback | WARNING[23385] chan_dahdi.c: Unrecognized prilocaldialplan NPI modifier: l |
| 19:43.07 | windback | it seems that using letters in sip users give this problem when calling over DHADI channels |
| 19:43.11 | windback | any ideas? |
| 19:44.06 | [TK]D-Fender | windback: "Using letters"? Pardon? |
| 19:44.13 | Qwell | windback: Show us the Dial() line |
| 19:44.34 | [TK]D-Fender | windback: Show us the full call with PRI debug as well |
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| 19:46.28 | nny | [TK]D-Fender: you're a gentleman and a scholar as always, both issues resolved. Cheers |
| 19:46.58 | [TK]D-Fender | nny: My favourite : You are good and kind. Good for nothing and kinda funny lookin' ;) |
| 19:47.27 | nny | [TK]D-Fender: lol i will remember that. Just mentally replace what I said with that |
| 19:48.39 | windback | [TK]D-Fender: http://pastebin.ca/1954272 |
| 19:48.49 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:49.46 | [TK]D-Fender | windback: I said PRI DEBUG |
| 19:50.06 | windback | [TK]D-Fender: Ok will do a call with pri debug enabled |
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| 19:56.54 | windback | [TK]D-Fender: http://pastebin.ca/1954277 |
| 19:57.12 | windback | (with pri debug enabled) |
| 19:58.46 | [TK]D-Fender | windback: those alpha chars part of the callerId NUMBER? |
| 19:59.04 | windback | [TK]D-Fender: yes |
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| 20:00.54 | [TK]D-Fender | windback: that is in indeed bad. |
| 20:01.27 | [TK]D-Fender | windback: Doesn't seem to stop your system from dialing out with partial callerid. Of course you should probably be SETTING it to something more meaningful. |
| 20:02.34 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 20:03.06 | windback | [TK]D-Fender: My sip extensions is called gyl536. Is that wrong? |
| 20:03.27 | windback | [TK]D-Fender: or should i change the callerid before calling over dahdi? |
| 20:03.48 | [TK]D-Fender | windback: Yes |
| 20:04.10 | windback | I should change the caller id before calling? |
| 20:06.21 | windback | [TK]D-Fender: yes for the first thing: I can't use alpha characters in sip extensions or yes for the second: I should change the sip caller id before calling over dahdi channel? |
| 20:06.35 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:07.50 | [TK]D-Fender | windback: Yes, you should not be sending alpha chars as the NUMBEr to a PRI |
| 20:08.41 | windback | [TK]D-Fender: which is the best way you consider I can change this? |
| 20:09.28 | [TK]D-Fender | windback: There is only 1 way to do this. "core show function CALLERID" |
| 20:19.27 | windback | [TK]D-Fender: thank. I though that perhaps I can change it from sip.conf using callerid=xxxx |
| 20:19.56 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 20:19.59 | wcselby | o/ |
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| 20:20.54 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 20:21.20 | ruben23 | hi guys any idea on this error on my CLI-->http://pastebin.com/cQM65RuR |
| 20:23.33 | ruben23 | guys any idea on this error |
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| 20:24.53 | wcselby | ruben23 - i have an idea, but I would need more information from you to know if my idea is related to your issue |
| 20:25.07 | wcselby | when does this error pop up |
| 20:25.17 | wcselby | what is happening, etc. how is your system setup? |
| 20:29.28 | ruben23 | <PROTECTED> |
| 20:29.53 | wcselby | dialing how? |
| 20:30.06 | wcselby | itsp, internal, analog, digital, help us help you |
| 20:33.08 | ruben23 | wcselby: voip |
| 20:33.21 | ruben23 | through SIP- softphones |
| 20:34.51 | wcselby | you're not helping |
| 20:35.32 | ruben23 | wcselby:uisng ITSO also-voip-softphones SIP. what you need else..? |
| 20:36.21 | ruben23 | wcselby: what do you want more..? |
| 20:37.55 | wcselby | i don't even understand what's happening in any kind of detail when you're getting this error message. maybe it's the language barrier. I don't know what "some is dialing my asterisk server" means. explain to me, in detail, what's happening when this error appears. show me relevant config files. maybe a sip debug. help us help you. |
| 20:41.02 | nextime | wcselby : is asterisk configured to allow only g729? the device is sending with another codec |
| 20:41.10 | nextime | https://issues.asterisk.org/view.php?id=9445 |
| 20:41.23 | nextime | can be the same issue? |
| 20:41.40 | nextime | ops |
| 20:41.45 | nextime | it was for ruben23 |
| 20:42.47 | nextime | ( i've just searched for the error on google... ) |
| 20:49.28 | pabelanger | Any recommendations for perl or php automated dialer? |
| 20:53.43 | ruben23 | wcselby: sorry, yes dialing through softphones using SIP going to US numbers, using g729 codec.. |
| 20:57.19 | bbryant | ~help |
| 20:58.09 | bbryant | what's the command for a list of US sip providers? |
| 20:58.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:59.59 | p3nguin_ | ~itsp |
| 21:00.00 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:00.10 | bbryant | ~itsplist-us |
| 21:00.10 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:00.19 | bbryant | thanks, p3nguin_ |
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| 21:10.31 | leifmadsen | pabelanger: ya, don't build one :D |
| 21:10.36 | leifmadsen | ducks and hides |
| 21:10.54 | p3nguin_ | I did it with bash. |
| 21:13.01 | pabelanger | leifmadsen: Client needs one, nothing special. I was trying to see if there we any existing 'scripts' already created to poll a database, check spool, then initial Originate(), guess not. Shouldn't be too hard to toss something together |
| 21:14.15 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 21:14.43 | raden_work | If i have a table of phone numbers in mysql is there a way i can auto dial off that table ? |
| 21:16.02 | pabelanger | raden_work: No, not directly from Asterisk. You'll need a script / application to tell Asterisk to dial them |
| 21:16.30 | raden_work | Interesting ... . |
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| 21:26.45 | bbryant | raden_work: you could create call files from SQL only, but after that you'd have to move them to spool using an external script of some sort |
| 21:27.07 | bbryant | something like "mv $mydir/*.call /var/lib/asterisk/outgoing/spool/" |
| 21:27.46 | bbryant | you could do it in a bash script |
| 21:28.13 | bbryant | raden_work: if you're interested in the SQL method lookup "INTO DUMPFILE" |
| 21:35.23 | bbryant | you'd need dialplan foo as well, but that would get it to dial them |
| 21:38.43 | raden_work | bbryant, I know php pretty well |
| 21:39.24 | raden_work | we have seminars and like have 6000 people in our customer file and would just like to send out friendly invites and people just cant do it fast enough and were always behind the 8 ball |
| 21:42.00 | fifer | How can you verif that a given dahdi PRI span is usign the timing of the far end? |
| 21:42.35 | fifer | I apear to have some timing issues that I'm trying to verify and fix. This is mainly effecting fax reception. |
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| 21:42.42 | Godfather_ | o/ |
| 21:42.49 | WIMPy | fifer: dmesg |
| 21:43.12 | *** part/#asterisk clintc (~clintc@n128-227-99-133.xlate.ufl.edu) |
| 21:46.25 | fifer | What am I looking for? I see all the "buffer re-sync occur from x to x" issues but I believe most if not all of those are related to my fxs card, not the dual T1 card. I see a "buffer sync missed!" message every once and a while |
| 21:46.42 | bbryant | raden_work: I used to do a web dev job in php not too long ago |
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| 21:47.04 | raden_work | bbryant, think i can do my auto dial with it ? |
| 21:47.29 | bbryant | yep. |
| 21:48.00 | bbryant | raden_work: http://phpagi.sourceforge.net/ |
| 21:48.18 | bbryant | disclaimer: I haven't used that, but that's what I found when googling "php" and "agi" |
| 21:48.25 | bbryant | which are the two things you'd need to accomplish your mission |
| 21:49.05 | WIMPy | fifer: When dahdi is started it tells you what timing source it's using. There are some fake ones at the very begining though. |
| 21:49.43 | fifer | ok, I'll look for the last system start, just a few days ago |
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| 22:03.35 | leifmadsen | pabelanger: ya sorry I wasn't trying to be dick :) I should have just answered better and said, "Not that I'm aware of." |
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| 23:03.37 | citywok | I have a weird CDR issue, and it's probably just my (mis)understanding. http://pastebin.com/KnDQpxS0 -- call comes in to agent, agent opens line 2 on phone and transfers the call to another agent. for some reason 2 entries get created for the second elg of the call... take a look at the pastebin. |
| 23:09.50 | pabelanger | leifmadsen: All cool |
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